TELECOM WORLD

SAMSUNG GALAXY S3

2012-06-06T04:08:00.000-07:00

HANDOVER EXPLAINED

2012-06-06T04:01:00.001-07:00

A hard handover is one in which the channel in the source cell is released and only then the channel in the target cell is engaged. Thus the connection to the source is broken before or 'as' the connection to the target is made—for this reason such handovers are also known as break-before-make. Hard handovers are intended to be instantaneous in order to minimize the disruption to the call. A hard handover is perceived by network engineers as an event during the call. It requires the least processing by the network providing service. When the mobile is between base stations, then the mobile can switch with any of the base stations, so the base stations bounce the link with the mobile back and forth. This is called ping-ponging.

A soft handover is one in which the channel in the source cell is retained and used for a while in parallel with the channel in the target cell. In this case the connection to the target is established before the connection to the source is broken, hence this handover is called make-before-break. The interval, during which the two connections are used in parallel, may be brief or substantial. For this reason the soft handover is perceived by network engineers as a state of the call, rather than a brief event. Soft handovers may involve using connections to more than two cells: connections to three, four or more cells can be maintained by one phone at the same time. When a call is in a state of soft handover, the signal of the best of all used channels can be used for the call at a given moment or all the signals can be combined to produce a clearer copy of the signal. The latter is more advantageous, and when such combining is performed both in the downlink (forward link) and the uplink (reverse link) the handover is termed as softer. Softer handovers are possible when the cells involved in the handovers have a single cell site.

An advantage of the hard handover is that at any moment in time one call uses only one channel. The hard handover event is indeed very short and usually is not perceptible by the user. In the old analog systems it could be heard as a click or a very short beep, in digital systems it is unnoticeable. Another advantage of the hard handoff is that the phone's hardware does not need to be capable of receiving two or more channels in parallel, which makes it cheaper and simpler. A disadvantage is that if a handover fails the call may be temporarily disrupted or even terminated abnormally. Technologies, which use hard handovers, usually have procedures which can re-establish the connection to the source cell if the connection to the target cell cannot be made. However re-establishing this connection may not always be possible (in which case the call will be terminated) and even when possible the procedure may cause a temporary interruption to the call.

One advantage of the soft handovers is that the connection to the source cell is broken only when a reliable connection to the target cell has been established and therefore the chances that the call will be terminated abnormally due to failed handovers are lower. However, by far a bigger advantage comes from the mere fact that simultaneously channels in multiple cells are maintained and the call could only fail if all of the channels are interfered or fade at the same time. Fading and interference in different channels are unrelated and therefore the probability of them taking place at the same moment in all channels is very low. Thus the reliability of the connection becomes higher when the call is in a soft handover. Because in a cellular network the majority of the handovers occur in places of poor coverage, where calls would frequently become unreliable when their channel is interfered or fading, soft handovers bring a significant improvement to the reliability of the calls in these places by making the interference or the fading in a single channel not critical. This advantage comes at the cost of more complex hardware in the phone, which must be capable of processing several channels in parallel. Another price to pay for soft handovers is use of several channels in the network to support just a single call. This reduces the number of remaining free channels and thus reduces the capacity of the network. By adjusting the duration of soft handovers and the size of the areas, in which they occur, the network engineers can balance the benefit of extra call reliability against the price of reduced capacity.

GBR IN LTE

2012-05-01T06:21:00.000-07:00

GBR

Guaranteed Bit Rate. A minimum bit rate requested by an application. In LTE, minimum GBR bearers and non-GBR bearers may be provided. Minimum GBR bearers are typically used for applications like Voice over Internet Pro- tocol (VoIP), with an associated GBR value; higher bit rates can be allowed if resources are available. Non-GBR bearers do not guarantee any particular bit rate, and are typically used for applications as web-browsing.

There are currently no posts in this category.

AMSUNG GALAXY S PLUS SPECIFICAYIONS

2012-04-02T03:06:00.001-07:00

ENERAL	2G Network	GSM 850 / 900 / 1800 / 1900
	3G Network	HSDPA 900 / 1900 / 2100
	Announced	2011, April
	Status	Available. Released 2011, July

BODY	Dimensions	122.4 x 64.2 x 9.9 mm
	Weight	119 g
		- Touch-sensitive controls

DISPLAY	Type	Super AMOLED capacitive touchscreen, 16M colors
	Size	480 x 800 pixels, 4.0 inches (~233 ppi pixel density)
	Multitouch	Yes
	Protection	Corning Gorilla Glass
		- TouchWiz 3.0 UI

SOUND	Alert types	Vibration; MP3, WAV ringtones
	Loudspeaker	Yes
	3.5mm jack	Yes

MEMORY	Card slot	microSD, up to 32GB
MEMORY	Internal	8/16 GB storage (market dependent), 512 MB RAM, 512 MB ROM

DATA	GPRS	Class 12 (4+1/3+2/2+3/1+4 slots), 32 - 48 kbps
	EDGE	Class 12
	Speed	HSDPA, 14.4 Mbps; HSUPA, 5.76 Mbps
	WLAN	Wi-Fi 802.11 b/g/n, DLNA, Wi-Fi hotspot
	Bluetooth	Yes, v3.0 with A2DP
	USB	Yes, microUSB v2.0

CAMERA	Primary	5 MP, 2592 x 1944 pixels, autofocus,
	Features	Geo-tagging, touch focus, face and smile detection
	Video	Yes, 720p@30fps
	Secondary	Yes, VGA

FEATURES	OS	Android OS, v2.3 (Gingerbread)
	Chipset	Qualcomm MSM8255T Snapdragon
	CPU	1.4 GHz Scorpion
	GPU	Adreno 205
	Sensors	Accelerometer, proximity, compass
	Messaging	SMS(threaded view), MMS, Email, Push Mail, IM, RSS
	Browser	HTML, Adobe Flash
	Radio	Stereo FM radio with RDS
	GPS	Yes, with A-GPS support
	Java	Yes, via Java MIDP emulator
	Colors	Black, White
		- Social networking integration - MP4/DivX/Xvid/WMV/H.264/H.263 player - MP3/WAV/eAAC+/AC3/FLAC player - TV-out - Organizer - Image/video editor - Document editor (Word, Excel, PowerPoint, PDF) - Google Search, Maps, Gmail, YouTube, Calendar, Google Talk, Picasa integration - Voice memo/dial/commands - Predictive text input (Swype)

BATTERY		Standard battery, Li-Ion 1650 mAh
	Stand-by	Up to 480 h (2G) / Up to 430 h (3G)
	Talk time	Up to 17 h 20 min (2G) / Up to 7 h 50 min (3G)

MISC	SAR US	0.44 W/kg (head) 0.94 W/kg (body)
	SAR EU	0.35 W/kg (head)
	Price group

SAMSUNG GALAXY S PLUS

2012-04-02T03:02:00.001-07:00

SAMSUNG GALAXY NEXUS SPECIFICATIONS

2012-04-02T03:00:00.001-07:00

GENERAL	2G Network	GSM 850 / 900 / 1800 / 1900
	3G Network	HSDPA 850 / 900 / 1700 / 1900 / 2100
	Announced	2011, October
	Status	Available. Released 2011, November

BODY	Dimensions	135.5 x 67.9 x 8.9 mm
BODY	Weight	135 g

DISPLAY	Type	Super AMOLED capacitive touchscreen, 16M colors
	Size	720 x 1280 pixels, 4.65 inches (~316 ppi pixel density)
	Multitouch	Yes
	Protection	Oleophobic coating

SOUND	Alert types	Vibration; MP3, WAV ringtones
	Loudspeaker	Yes
	3.5mm jack	Yes, check quality

MEMORY	Card slot	No
MEMORY	Internal	16 GB storage, 1 GB RAM

DATA	GPRS	Yes
	EDGE	Yes
	Speed	HSDPA, 21 Mbps; HSUPA, 5.76 Mbps
	WLAN	Wi-Fi 802.11 a/b/g/n, dual-band, DLNA, Wi-Fi hotspot
	Bluetooth	Yes, v3.0 with A2DP
	NFC	Yes
	USB	Yes, microUSB v2.0 (MHL)

CAMERA	Primary	5 MP, 2592x1944 pixels, autofocus, LED flash, check quality
	Features	Touch focus, geo-tagging, face detection
	Video	Yes, 1080p@30fps, check quality
	Secondary	Yes, 1.3 MP, 720p@30fps

FEATURES	OS	Android OS, v4.0 (Ice Cream Sandwich), upgradable to v4.0.4
	Chipset	TI OMAP 4460
	CPU	Dual-core 1.2 GHz Cortex-A9
	GPU	PowerVR SGX540
	Sensors	Accelerometer, gyro, proximity, compass, barometer
	Messaging	SMS(threaded view), MMS, Email, Push Mail, IM, RSS
	Browser	HTML, Adobe Flash
	Radio	No
	GPS	Yes, with A-GPS support
	Java	Yes, via Java MIDP emulator
	Colors	Black, White
		- Active noise cancellation with dedicated mic - TV-out (via MHL A/V link) - MP4/H.264/H.263 player - MP3/WAV/eAAC+/AC3 player - Organizer - Image/video editor - Document viewer - Google Search, Maps, Gmail, YouTube, Calendar, Google Talk, Picasa integration - Voice memo/dial/commands - Predictive text input

BATTERY		Standard battery, Li-Ion 1750 mAh
	Stand-by	Up to 290 h (2G) / Up to 270 h (3G)
	Talk time	Up to 17 h 40 min (2G) / Up to 8 h 20 min (3G)

MISC	SAR EU	0.30 W/kg (head)
MISC	Price group

SAMSUNG GALAXY NEXUS

2012-04-02T02:59:00.001-07:00

what is fading , inter symbol interference???

2012-03-29T12:11:00.002-07:00

1.2.1 Attenuation

Attenuation is the drop in the signal power when transmitting from one point to another. It can be caused by the transmission path length, obstructions in the signal path, and multipath effects. Figure 2 shows some of the radio propagation effects that cause attenuation. Any objects that obstruct the line of sight signal from the transmitter to the receiver can cause attenuation.

Figure 2 Radio Propagation Effects

Shadowing of the signal can occur whenever there is an obstruction between the transmitter and receiver. It is generally caused by buildings and hills, and is the most important environmental attenuation factor.

Shadowing is most severe in heavily built up areas, due to the shadowing from buildings. However, hills can cause a large problem due to the large shadow they produce. Radio signals diffract off the boundaries of obstructions, thus preventing total shadowing of the signals behind hills and buildings. However, the amount of diffraction is dependent on the radio frequency used, with low frequencies diffracting more then high frequency signals. Thus high frequency signals, especially, Ultra High Frequencies (UHF), and microwave signals require line of sight for adequate signal strength. To over come the problem of shadowing, transmitters are usually elevated as high as possible to minimise the number of obstructions. Typical amounts of variation in attenuation due to shadowing are shown in Table 6.

Description	Typical Attenuation due to Shadowing
Heavily built-up urban centre	20dB variation from street to street
Sub-urban area (fewer large buildings)	10dB greater signal power then built-up urban center
Open rural area	20dB greater signal power then sub-urban areas
Terrain irregularities and tree foliage	3-12dB signal power variation

Table 6 Typical shadowing in a radio channel (Values from _[11])

Shadowed areas tend to be large, resulting in the rate of change of the signal power being slow. For this reason, it is termed slow-fading, or log-normal shadowing.

1.2.2 Multipath Effects

1.2.2.1 Rayleigh fading

In a radio link, the RF signal from the transmitter may be reflected from objects such as hills, buildings, or vehicles. This gives rise to multiple transmission paths at the receiver. Figure 3 show some of the possible ways in which multipath signals can occur.

Figure 3 Multipath Signals

The relative phase of multiple reflected signals can cause constructive or destructive interference at the receiver. This is experienced over very short distances (typically at half wavelength distances), thus is given the term fast fading. These variations can vary from 10-30dB over a short distance. Figure 4 shows the level of attenuation that can occur due to the fading.

Figure 4 Typical Rayleigh fading while the Mobile Unit is moving (for at 900 MHz)_[15]

The Rayleigh distribution is commonly used to describe the statistical time varying nature of the received signal power. It describes the probability of the signal level being received due to fading. Table 7 shows the probability of the signal level for the Rayleigh distribution.

Signal Level (dB about median)	% Probability of Signal Level being less then the value given
10	99
0	50
-10	5
-20	0.5
-30	0.05

Table 7 Cummulative distribution for Rayleigh distribution (Value from _[15])

1.2.2.2 Frequency Selective Fading

In any radio transmission, the channel spectral response is not flat. It has dips or fades in the response due to reflections causing cancellation of certain frequencies at the receiver. Reflections off near-by objects (e.g. ground, buildings, trees, etc) can lead to multipath signals of similar signal power as the direct signal. This can result in deep nulls in the received signal power due to destructive interference.

For narrow bandwidth transmissions if the null in the frequency response occurs at the transmission frequency then the entire signal can be lost. This can be partly overcome in two ways.

By transmitting a wide bandwidth signal or spread spectrum as CDMA, any dips in the spectrum only result in a small loss of signal power, rather than a complete loss. Another method is to split the transmission up into many small bandwidth carriers, as is done in a COFDM/OFDM transmission. The original signal is spread over a wide bandwidth and so nulls in the spectrum are likely to only affect a small number of carriers rather than the entire signal. The information in the lost carriers can be recovered by using forward error correction techniques.

1.2.2.3 Delay Spread

The received radio signal from a transmitter consists of typically a direct signal, plus reflections off objects such as buildings, mountings, and other structures. The reflected signals arrive at a later time then the direct signal because of the extra path length, giving rise to a slightly different arrival times, spreading the received energy in time. Delay spread is the time spread between the arrival of the first and last significant multipath signal seen by the receiver.

In a digital system, the delay spread can lead to inter-symbol interference. This is due to the delayed multipath signal overlapping with the following symbols. This can cause significant errors in high bit rate systems, especially when using time division multiplexing (TDMA). Figure 5 shows the effect of inter-symbol interference due to delay spread on the received signal. As the transmitted bit rate is increased the amount of inter-symbol interference also increases. The effect starts to become very significant when the delay spread is greater then ~50% of the bit time.

Figure 5 Multipath Delay Spread

Table 8 shows the typical delay spread for various environments. The maximum delay spread in an outdoor environment is approximately 20 us, thus significant inter-symbol interference can occur at bit rates as low as 25 kbps.

Environment or cause	Delay Spread	Maximum Path Length Difference
Indoor (room)	40 nsec - 200 nsec	12 m - 60 m
Outdoor	1 m sec - 20 m sec	300 m - 6 km

Table 8 Typical Delay Spread

Inter-symbol interference can be minimized in several ways. One method is to reduce the symbol rate by reducing the data rate for each channel (i.e. split the bandwidth into more channels using frequency division multiplexing, or OFDM). Another is to use a coding scheme that is tolerant to inter-symbol interference such as CDMA.

WHAT IS CISC????

2012-03-20T07:14:00.002-07:00

Pronounced sisk, and stands for complex instruction set computer. Mostpersonal computers, use a CISC architecture, in which the CPU supports as many as two hundred instructions. An alternative architecture, used by manyworkstations and also some personal computers, is RISC (reduced instruction set computer), which supports fewer instructions.

WHAT IS RISC???

2012-03-20T07:13:00.001-07:00

Pronounced risk, acronym for reduced instruction set computer, a type ofmicroprocessor that recognizes a relatively limited number of instructions. Until the mid-1980s, the tendency among computer manufacturers was to build increasingly complex CPUs that had ever-larger sets of instructions. At that time, however, a number of computer manufacturers decided to reverse this trend by building CPUs capable of executing only a very limited set of instructions. One advantage of reduced instruction set computers is that they can execute their instructions very fast because the instructions are so simple. Another, perhaps more important advantage, is that RISC chipsrequire fewer transistors, which makes them cheaper to design and produce. Since the emergence of RISC computers, conventional computers have been referred to as CISCs (complex instruction set computers).

There is still considerable controversy among experts about the ultimate value of RISC architectures. Its proponents argue that RISC machines are both cheaper and faster, and are therefore the machines of the future. Skeptics note that by making the hardware simpler, RISC architectures put a greater burden on the software. They argue that this is not worth the trouble because conventional microprocessors are becoming increasingly fast and cheap anyway.

To some extent, the argument is becoming moot because CISC and RISC implementations are becoming more and more alike. Many of today's RISC chips support as many instructions as yesterday's CISC chips. And today's CISC chips use many techniques formerly associated with RIS

WHAT IS TRANSISTOR????

2012-03-20T07:08:00.002-07:00

A device composed of semiconductor material that amplifies a signal or opens or closes a circuit. Invented in 1947 at Bell Labs, transistors have become the key ingredient of all digital circuits, including computers. Today's microprocessors contains tens of millions of microscopic transistors.

Prior to the invention of transistors, digital circuits were composed of vacuum tubes, which had many disadvantages. They were much larger, required more energy, dissipated more heat, and were more prone to failures. It's safe to say that without the invention of transistors, computing as we know it today would not be possible.

WHAT IS GPU????

2012-03-20T07:05:00.000-07:00

Used primarily for 3-D applications, a graphics processing unit is a single-chip processor that creates lighting effects and transforms objects every time a 3D scene is redrawn. These are mathematically-intensive tasks, which otherwise, would put quite a strain on the CPU. Lifting this burden from the CPU frees up cycles that can be used for other jobs.

The first company to develop the GPU is NVIDIA Inc. Its GeForce 256 GPU is capable of billions of calculations per second, can process a minimum of 10 million polygons per second, and has over 22 million transistors, compared to the 9 million found on the Pentium III. Its workstation version called the Quadro, designed for CAD applications, can process over 200 billion operations a second and deliver up to 17 million triangles per second.

The GeForce NVIDIA GPU card is compatible with the following graphics APIs: OpenGL and Microsoft's DirectX, Intel's Accelerated Graphics Port (AGP) technology and AMD 's 3DNow!

The Quadro is an OpenGL specific card with driver support for Pentium III Xeon and AMD Athlon CPUs.

TDMA TECHNOLOGY EXPLAINED

2012-03-13T07:04:00.003-07:00

As you remember from the Introduction to TDMA tutorial. GSM divides up each ARFCN into 8 time slots.

These 8 timeslots are further broken up into logical channels.

Logical channels can be thought of as just different types of data that is transmitted only on certain frames in a certain timeslot.

Different time slots will carry different logical channels, depending on the structure the BSS uses.
There are two main categories of logical channels in GSM:

Signaling Channels
Traffic Channels (TCH)

Signaling Channels

These are the main types of signaling Channels:Broadcast Channels (BCH) - Transmitted by the BTS to the MS. This channel carries system parameters needed to identify the network, synchronize time and frequency with the network, and gain access to the network.

Common Control Channels (CCH) - Used for signaling between the BTS and the MS and to request and grant access to the network.

Standalone Dedicated Control Channels (SDCCH) - Used for call setup.

Associated Control Channels (ACCH) - Used for signaling associated with calls and call-setup. An ACCH is always allocated in conjunction with a TCH or a SDCCH.

*keep in mind, these are only categories of logical channels, they are not logical channels themselves.

The above categories can be divided into the following logical channels:

Broadcast Channels (BCH)
     Broadcast Control Channel (BCCH)
     Frequency Correction Channel (FCCH)
     Synchronization Channel (SCH)
     Cell Broadcast Channel (CBCH)

Common Control Channels (CCCH)
     Paging Channel (PCH)
     Random Access Channel (RACH)
     Access Grant Channel (AGCH)

Dedicated Control Channel (DCCH)
     Standalone Dedicated Control Channel (SDCCH)
     Fast Associated Control Channel (FACCH)
     Slow Associated Control Channel (SACCH)

Let's examine each type of logical channel individually.

Broadcast Channels (BCH)

Broadcast Control Channel (BCCH) - DOWNLINK - This channel contains system parameters needed to identify the network and gain access. These paramters include the Location Area Code (LAC), the Mobile Network Code (MNC), the frequencies of neighboring cells, and access parameters.

Frequency Correction Channel (FCCH) - DOWNLINK - This channel is used by the MS as a frequency reference. This channel contains frequency correction bursts.

Synchronization Channel (SCH) - DOWNLINK - This channel is used by the MS to learn the Base Station Information Code (BSIC) as well as the TDMA frame number (FN). This lets the MS know what TDMA frame they are on within the hyperframe.
* The BSIC was covered in the Introduction to GSM Tutorial. You can also read about the numbering schemes used in GSM.

Cell Broadcast Channel (CBCH) - DOWNLINK - This channel is not truly its own type of logical channel. The CBCH is for point-to-omnipoint messages. It is used to broadcast specific information to network subscribers; such as weather, traffic, sports, stocks, etc. Messages can be of any nature depending on what service is provided. Messages are normally public service type messages or announcements. The CBCH isnt allocated a slot for itself, it is assigned to an SDCCH. It only occurs on the downlink. The CBCH usually occupies the second subslot of the SDCCH. The mobile will not acknowledge any of the messages.

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Common Control Channels (CCCH)

Paging Channel (PCH) - DOWNLINK - This channel is used to inform the MS that it has incoming traffic. The traffic could be a voice call, SMS, or some other form of traffic.

Random Access Channel (RACH) - UPLINK This channel is used by a MS to request an initial dedicated channel from the BTS. This would be the first transmission made by a MS to access the network and request radio resources. The MS sends an Access Burst on this channel in order to request access.

Access Grant Channel (AGCH) - DOWNLINK - This channel is used by a BTS to notify the MS of the assignement of an initial SDCCH for initial signaling.

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Dedicated Control Channels (DCCH)

Standalone Dedicated Control Channel (SDCCH) - UPLINK/DOWNLINK - This channel is used for signaling and call setup between the MS and the BTS.

Fast Associated Control Channel (FACCH) - UPLINK/DOWNLINK - This channel is used for control requirements such as handoffs. There is no TS and frame allocation dedicated to a FAACH. The FAACH is a burst-stealing channel, it steals a Timeslot from a Traffic Channel (TCH).

Slow Associated Control Channel (SACCH) - UPLINK/DOWNLINK - This channel is a continuous stream channel that is used for control and supervisory signals associated with the traffic channels.

Signaling Channel Mapping

Normally the first two timeslots are allocated to signaling channels.

Remember that Control Channel (aka signaling channels) are composed of 51 TDMA frames. On a time slot Within the multiframe, the 51 TDMA frames are divided up and allocated to the various logical channels.

There are several channel combinations allowed in GSM. Some of the more common ones are:
FCCH + SCH + BCCH + CCCH
BCCH + CCCH
FCCH + SCH + BCCH + CCCH + SDCCH/4(0..3) + SACCH/C4(0..3)
SDCCH/8(0 .7) + SACCH/C8(0 . 7)

FCCH + SCH + BCCH + CCCH

Downlink

Uplink

[

BCCH + CCCH

Downlink

Uplink

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FCCH + SCH + BCCH + CCCH + SDCCH/4(0..3) + SACCH/C4(0..3)

The SACCH that is associated with each SDCCH is only transmitted every other multiframe. Each SACCH only gets half of the transmit time as the SDCCH that it is associated with. So, in one multiframe, SACCH0 and SACCH1 would be transmitted, and in the next multiframe, SACCH2 and SACCH3 would be transmitted. The two sequential multiframes would look like this:

Downlink

Uplink

You will also notice that the downlink and uplink multiframes do not align with each other. This is done so that if the BTS sends an information request to the MS, it does not have to wait an entire multiframes to receive the needed information. The uplink is transmitted 15 TDMA frames behind the downlink. For example, the BTS might send an authentication request to the MS on SDCCH0 (downlink) which corresponds to TDMA frames 22-25. The MS then has enough time to process the request and reply on SDCCH0 (uplink) which immediately follows it on TDMA frames 37-40.

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SDCCH/8(0 .7) + SACCH/C8(0 . 7)

Once again, the SACCH that is associated with an SDCCH is only transmitted every other multiframe. Two consecutive multiframes would look like this:

Downlink

Uplink

Traffic Channels (TCH)

Traffic Channels are used to carry two types of information to and from the user:

Encoded Speech
Data

There are two basic types of Encoded Speech channels:

Encoded Speech - Encoded speech is voice audio that is converted into digital form and compressed. See the Speech Encoding tutorial to see the process.
    Full Rate Speech TCH (TCH/FS) - 13 kb/s
    Half Rate Speech TCH (TCH/HS) - 5.6 kb/s

Data - Data refers to user data such as text messages, picture messages, internet browsing, etc. It includes pretty much everything except speech.

    Full rate Data TCH (TCH/F14.1) - 14.4 kb/s
    Full rate Data TCH (TCH/F9.6) - 9.6 kb/s
    Full rate Data TCH (TCH/F4.8) - 4.8 kb/s
    Half rate Data TCH (TCH/F4.8) - 4.8 kb/s
    Full rate Data TCH (TCH/F2.4) - ≤2.4 kb/s
    Half rate Data TCH (TCH/H2.4) - ≤2.4 kb/s

Traffic Channel Mapping

Time slots 2 through 7 are normally used for Traffic Channels (TCH)

Traffic Channel Multiframes are composed of only 26 TDMA frames. On each multiframe, there are 24 frames for Traffic Channels, 1 frame for a SACCH, and the last frame is Idle. Remember that a MS (or other device) only gets one time slot per TDMA frame to transmit, so in the following diagrams we are looking at a single time slot.

Full Rate Traffic Channel (TCH/FS)

When using Half-Rate Speech Encoding (TCH/HS), the speech encoding bit rate is 5.6 kb/s, so one time slot can handle two half-rate channels. In this case, one channel will transmit every other TDMA frame, and the other channel would be transmitted on the other frames. The final frame (25), which is normally used as an Idle frame, is now used as a SACCH for the second half-rate channel.

Half Rate Traffic Channel (TCH/HS)

ARFCN Mapping

This diagram shows a sample Multiframe with logical channels mapped to time slots and TDMA frames. This is just one possible configuration for an ARFCN.
*For illustrative purposes, half of the traffic channels are full-rate and the other half are half-rate

TS0
TS1
TS2
TS3
TS4
TS5
TS6
TS7

SIGNALLING SYSTEM 7 (SS 7) EXPLAINED

2012-03-13T01:46:00.003-07:00

Signaling System No. 7 (SS7) is a set of telephony signaling protocols which are used to set up most of the world's public switched telephone network telephone calls. The main purpose is to set up andtear down telephone calls. Other uses include number translation, prepaid billing mechanisms, short message service (SMS), and a variety of other mass market services.

It is usually referenced as Signaling System No. 7 or Signaling System #7, or simply abbreviated to SS7. In North America it is often referred to as CCSS7, an acronym for Common Channel Signaling System 7. In some European countries, specifically the United Kingdom, it is sometimes called C7 (CCITT number 7) and is also known as number 7 and CCIS7 (Common Channel Interoffice Signaling 7).

There is only one international SS7 protocol defined by ITU-T in its Q.700-series recommendations. There are however, many national variants of the SS7 protocols. Most national variants are based on two widely deployed national variants as standardized by ANSI and ETSI, which are in turn based on the international protocol defined by ITU-T. Each national variant has its own unique characteristics. Some national variants with rather striking characteristics are the China (PRC) and Japan (TTC) national variants.

The Internet Engineering Task Force (IETF) has also defined level 2, 3, and 4 protocols that are compatible with SS7:

§ MTP2 (M2UA and M2PA)

§ MTP3 (M3UA)

§ Signalling Connection Control Part (SCCP) (SUA)

but use a Stream Control Transmission Protocol (SCTP) transport mechanism. This suite of protocols is called SIGTRAN

There are two essential components to all telephone calls. The first, and most obvious, is the actual content—our voices, faxes, modem data, etc. The second is the information that instructs telephone exchanges to establish connections and route the “content” to an appropriate destination. Telephony signaling is concerned with the creation of standards for the latter to achieve the former. These standards are known as protocols. SS7 or Signaling System Number 7 is simply another set of protocols that describe a means of communication between telephone switches in public telephone networks. They have been created and controlled by various bodies around the world, which leads to some specific local variations, but the principal organization with responsibility for their administration is the International Telecommunications Union or ITU-T.

Signalling System Number 7 (SS#7 or C7) is the protocol used by the telephone companies for interoffice signalling. In the past, in-band signalling techniques were used on interoffice trunks. This method of signalling used the same physical path for both the call-control signalling and the actual connected call. This method of signalling is inefficient and is rapidly being replaced by out-of-band or common-channel signalling techniques.

To understand SS7 we must first understand something of the basic inefficiency of previous signaling methods utilized in the Public Switched Telephone Network (PSTN). Until relatively recently, all telephone connections were managed by a variety of techniques centered on “in band” signaling.

A network utilizing common-channel signalling is actually two networks in one:

1. First there is the circuit-switched "user" network which actually carries the user voice and data traffic. It provides a physical path between the source and destination.

2. The second is the signalling network which carries the call control traffic. It is a packet-switched network using a common channel switching protocol.

SS7 PROTOCOL SUIT

The SS7 network is an interconnected set of network elements that is used to exchange messages in support of telecommunications functions. The SS7 protocol is designed to both facilitate these functions and to maintain the network over which they are provided. Like most modern protocols, the SS7 protocol is layered.

Physical Layer (MTP-1)
This defines the physical and electrical characteristics of the signaling links of the SS7 network. Signaling links utilize DS–0 channels and carry raw signaling data at a rate of 56 kbps or 64 kbps (56 kbps is the more common implementation).
Message Transfer Part—Level 2 (MTP-2)
The level 2 portion of the message transfer part (MTP Level 2) provides link-layer functionality. It ensures that the two end points of a signaling link can reliably exchange signaling messages. It incorporates such capabilities as error checking, flow control, and sequence checking.
Message Transfer Part—Level 3 (MTP-3)
The level 3 portion of the message transfer part (MTP Level 3) extends the functionality provided by MTP level 2 to provide network layer functionality. It ensures that messages can be delivered between signaling points across the SS7 network regardless of whether they are directly connected. It includes such capabilities as node addressing, routing, alternate routing, and congestion control.

Signaling Connection Control Part (SCCP)

The signaling connection control part (SCCP) provides two major functions that are lacking in the MTP. The first of these is the capability to address applications within a signaling point. The MTP can only receive and deliver messages from a node as a whole; it does not deal with software applications within a node.

While MTP network-management messages and basic call-setup messages are addressed to a node as a whole, other messages are used by separate applications (referred to as subsystems) within a node. Examples of subsystems are 800 call processing, calling-card processing, advanced intelligent network (AIN), and custom local-area signaling services (CLASS) services (e.g., repeat dialing and call return). The SCCP allows these subsystems to be addressed explicitly.

ISDN User Part (ISUP)

ISUP user part defines the messages and protocol used in the establishment and tear down of voice and data calls over the public switched network (PSN), and to manage the trunk network on which they rely. Despite its name, ISUP is used for both ISDN and non–ISDN calls. In the North American version of SS7, ISUP messages rely exclusively on MTP to transport messages between concerned nodes.

Transaction Capabilities Application Part (TCAP)

TCAP defines the messages and protocol used to communicate between applications (deployed as subsystems) in nodes. It is used for database services such as calling card, 800, and AIN as well as switch-to-switch services including repeat dialing and call return. Because TCAP messages must be delivered to individual applications within the nodes they address, they use the SCCP for transport.

Operations, Maintenance, and Administration Part (OMAP)

OMAP defines messages and protocol designed to assist administrators of the SS7 network. To date, the most fully developed and deployed of these capabilities are procedures for validating network routing tables and for diagnosing link troubles. OMAP includes messages that use both the MTP and SCCP for routing.

Signaling Elements/Endpoints

· SSPs are end office or tandem switches that connect voice circuits and perform the necessary signaling functions to originate and terminate calls.

· The STP routes all the signaling messages in the SS7 network.

· The SCP provides access to databases for additional routing information used in call processing. Also, the SCP is the key element for delivering IN applications on the telephony network.

The following sections explore the three signaling elements of the SS7 network in more detail.

SSP

SSPs are telephone switches that are provisioned with SS7 capabilities. End office SSPs originate and terminate calls, and core network switches (STPs) provide tandem or transit calls. The SSP provides circuit-based signaling messages to other SSPs for the purposes of connecting, disconnecting, and managing voice calls. Non-circuit based messages are used to query databases when the dialed number is insufficient to complete the call.

End office SSPs connect directly to users on their subscriber interfaces. The protocols used can vary from analog to digital and can be based on ISDN Primary Rate Interface (PRI) or channel-associated switching (CAS). The end office is in charge of translating subscriber protocol requests into SS7 messages to establish calls.

The SSP uses the dialed number to complete the call, unless, for example, it is an 800, 8xx, 9xx, or Local Number Portability exchange (or is ported NXX). In the latter case, a query is sent to an SCP requesting the routing information (number) necessary to complete the call.


1.	The SSP uses the called number from the calling party or routing number from the database query to begin circuit connection signaling messages.
2.	Then the SSP uses its routing table to determine the trunk group and circuit needed to connect the call.
3.	At this point, a signaling setup message is sent to the destination SSP requesting a connection on the circuit specified by the originating SSP.
4.	The destination SSP responds with an acknowledgment granting permission to connect to the specified trunk and proceeds to connect the call to the final destination.

STP

STPs, as illustrated, are an integral part of the SS7 architecture providing access to the network. STPs route or switch all the signaling messages in the network based on the routing information and destination point code address contained in the message.

The STP provides the logical connectivity between SSPs without requiring direct SSP-to-SSP links. STPs are configured in pairs and are mated to provide redundancy and higher availability. These mated STPs perform identical functions and are considered the home STPs for the directly connected SSP or SCP. The STP also is capable of performing global title translation, which is discussed later in this section.

Circuit-based messages are created on the SSP. Then, they are packetized in SS7 packets and sent from the SSP. Usually they contain requests to connect or disconnect a call. These packets are forwarded to the destination SSP where the call is terminated. It is the STP network's job to properly route such packets to the destination.

Non-circuit based messages that originate from an SSP are database queries requesting additional information needed to complete the call. It is the STP network's job to properly route packets between the SSP and the database interface known as the SCP. These packets are routed to the destination SCP and are addressed to the appropriate subsystem database. The SCP is the interface to the database that provides the routing number required to complete the call.

STPs also measure traffic and usage. Traffic measurements provide statistics such as network events and message types, and usage measurements provide statistics on the access and number of messages per message type. This information is used by the carrier's network planning teams to look at overall system capacity and future planning.

Global Title Translation

In addition to performing basic SS7 packet routing, STPs are capable of performing gateway services such as global title translation. This function is used to centralize the SCP and database selection versus distributing all possible destination selections to hundreds or thousands of distributed switches. If the SSP is unaware of the destination SCP address, it can send the database query to its local STP. The STP then performs global title translation and re-addresses the destination of the database query to the appropriate SCP.

Global title translation centralizes the selection of the correct database by enabling queries to be addressed directly to the STP. SSPs, therefore, do not have the burden of maintaining every potential destination database address. The term global title translation is taken from the term global title digits, which is another term for dialed digits.

The STP looks at the global dialed digits and through its own translation table to resolve the following:

· The point code address of the appropriate SCP for the database

· The subsystem number of the database

The STP also can perform an intermediate global title translation by using its translation table to find another STP. The intermediate STP then routes the message to the other STP to perform the final global title translation.

STP Hierarchy

STP hierarchy defines network interconnection and separates capabilities into specific areas of functionality. STP implementation can occur in multiple levels, such as:

· Local Signal Transfer Point

· Regional Signal Transfer Point

· National Signal Transfer Point

· International Signal Transfer Point

· Gateway Signal Transfer Point

The local, regional, and national STPs transfer standards-based SS7 messages within the same network. These STPs usually are not capable of converting or handling messages in different formats or versions.

International STPs provide international connectivity where the same International Telecommunication Union (ITU) standards are deployed in both networks.

Gateway STPs can provide the following:

· Protocol conversion from national versions to the ITU standard

· Network-to-network interconnection points

· Network security features such as screening, which is used to examine all incoming and outgoing messages to ensure authorization

You can deploy and install STP functions on separate dedicated devices or incorporate them with other SSP functions onto a single end office or tandem switch. Integrating SSP and STP functions is particularly common in Europe and Australia. This is why fully associated SS7 or CCS7 (CCS7 is the ITU-T version of SS7) networks are prevalent in those areas. Fully associated SS7 occurs when the same transmission channel carries the bearer's information and the signaling information.

SCP

The SCP, as shown provides the interface to the database where additional routing information is stored for non-circuit based messages. Service-provider SCPs do not house the required information; they do, however, provide the interface to the system's database. The interface between the SCP and the database system is accomplished by a standard protocol, which is typically X.25. The SCP provides the conversion between the SS7 and the X.25 protocol. If X.25 is not the database access protocol, the SCP still provides the capability for communication through the use of primitives.

The database stores information related to its application and is addressed by a subsystem number, which is unique for each database. The subsystem number is known at the SSP level; the request originated within the PSTN contains that identifier. The subsystem number identifies the database where the information is stored and is used by the SCP to respond to the request.

The following databases are the most common in the SS7 network:

· 800 DatabaseProvides the routing information for special numbers, such as 800, 877, 888, 900, and 976 numbers. The 800 database responds to the special number queries with the corresponding routing number. In the case of 800, 888, and 900 numbers, the routing number is the actual telephone number at the terminating end.

· Line Information Database (LIDB)Provides subscriber or user information such as screening and barring, calling-card services including card validation and personal identification number (PIN) authentication, and billing. The billing features of this database determine ways you can bill collect calls, calling-card calls, and third-party services.

· Local Number Portability Database (LNPDB)Provides the 10-digit Location Routing Number (LRN) of the switch that serves the dialed-party number. The LRN is used to route the call through the network, and the dialed-party number is used to complete the call at the terminating SSP.

· Home Location Register (HLR)Used in cellular networks to store information such as current cellular phone location, billing, and cellular subscriber information.

· Visitor Location Register (VLR)Used in cellular networks to store information on subscribers roaming outside the home network. The VLR uses this information to communicate to the HLR database to identify the subscriber's location when roaming.

VOIP TECHNOLOGY EXPLAINED

2012-03-12T11:30:00.002-07:00

Voice over Internet Protocol (Voice over IP, VoIP) is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. Other terms frequently encountered and often used synonymously with VoIP are IP telephony, Internet telephony, voice over broadband (VoBB), broadband telephony, and broadband phone.

Internet telephony refers to communications services—Voice, fax, SMS, and/or voice-messaging applications—that are transported via the Internet, rather than the public switched telephone network (PSTN). The steps involved in originating a VoIP telephone call are signaling and media channel setup, digitization of the analog voice signal, encoding, packetization, and transmission as Internet Protocol (IP) packets over a packet-switched network. On the receiving side, similar steps (usually in the reverse order) such as reception of the IP packets, decoding of the packets and digital-to-analog conversion reproduce the original voice stream. Even though IP Telephony and VoIP are terms that are used interchangeably, they are actually different; IP telephony has to do with digital telephony systems that use IP protocols for voice communication while VoIP is actually a subset of IP Telephony. VoIP is a technology used by IP telephony as a means of transporting phone calls.

VoIP systems employ session control protocols to control the set-up and tear-down of calls as well as audio codecs which encode speech allowing transmission over an IP network as digital audio via an audio stream. The codec used is varied between different implementations of VoIP (and often a range of codecs are used); some implementations rely on narrowband and compressed speech, while others support high fidelity stereo codecs.

There are three types of VoIP tools that are commonly used; IP Phones,Software VoIP and Mobile and Integrated VoIP. The IP Phones are the most institutionally established but still the least obvious of the VoIP tools. The use of software VoIP has increased during the global recession of 2008-2010, as many persons, looking for ways to cut costs have turned to these tools for free or inexpensive calling or video conferencing applications. Software VoIP can be further broken down into three classes or subcategories; Web Calling, Voice and Video Instant Messaging and Web Conferencing. Mobile and Integrated VoIP is just another example of the adaptability of VoIP. VoIP is available on many smartphones and internet devices so even the users of portable devices that are not phones can still make calls or send SMS text messages over 3G or WIFI.

PROTOCOLS

Voice over IP has been implemented in various ways using both proprietary and open protocols and standards. Examples of technologies used to implement Voice over IP include:

§ H.323

§ IP Multimedia Subsystem (IMS)

§ Media Gateway Control Protocol (MGCP)

§ Session Initiation Protocol (SIP)

§ Real-time Transport Protocol (RTP)

§ Session Description Protocol (SDP)

§ Inter-Asterisk eXchange (IAX)

The H.323 protocol was one of the first VoIP protocols that found widespread implementation for long-distance traffic, as well as local area network services. However, since the development of newer, less complex protocols, such as MGCP and SIP, H.323 deployments are increasingly limited to carrying existing long-haul network traffic. In particular, the Session Initiation Protocol (SIP) has gained widespread VoIP market penetration.

A notable proprietary implementation is the Skype protocol, which is in part based on the principles of Peer-to-Peer (P2P) networking.

IMPLEMENTATION

A major development that started in 2004^] was the introduction of mass-market VoIP services that utilize existing broadband Internet access, by which subscribers place and receive telephone calls in much the same manner as they would via the public switched telephone network (PSTN). Full-service VoIP phone companies provide inbound and outbound service with Direct Inbound Dialing. Many offer unlimited domestic calling for a flat monthly subscription fee. This sometimes includes international calls to certain countries. Phone calls between subscribers of the same provider are usually free when flat-fee service is not available.

A VoIP phone is necessary to connect to a VoIP service provider. This can be implemented in several ways:

§ Dedicated VoIP phones connect directly to the IP network using technologies such as wired Ethernet or wireless Wi-Fi. They are typically designed in the style of traditional digital business telephones.

§ An analog telephone adapter is a device that connects to the network and implements the electronics and firmware to operate a conventional analog telephone attached through a modular phone jack. Some residential Internet gateways and cablemodems have this function built in.

§ A softphone is application software installed on a networked computer that is equipped with a microphone and speaker, or headset. The application typically presents a dial pad and display field to the user to operate the application by mouse clicks or keyboard input.

CORPORATE USE

Because of the bandwidth efficiency and low costs that VoIP technology can provide, businesses are migrating from traditional copper-wire telephone systems to VoIP systems to reduce their monthly phone costs. In 2008, 80% of all new PBX lines installed internationally were VoIP.

VoIP solutions aimed at businesses have evolved into "unified communications" services that treat all communications—phone calls, faxes, voice mail, e-mail, Web conferences and more—as discrete units that can all be delivered via any means and to any handset, including cellphones. Two kinds of competitors are competing in this space: one set is focused on VoIP for medium to large enterprises, while another is targeting the small-to-medium business (SMB) market.

VoIP allows both voice and data communications to be run over a single network, which can significantly reduce infrastructure costs.

The prices of extensions on VoIP are lower than for PBX and key systems. VoIP switches may run on commodity hardware, such as PCs or Linux systems. Rather than closed architectures, these devices rely on standard interfaces.

VoIP devices have simple, intuitive user interfaces, so users can often make simple system configuration changes. Dual-mode cellphones enable users to continue their conversations as they move between an outside cellular service and an internal Wi-Finetwork, so that it is no longer necessary to carry both a desktop phone and a cellphone. Maintenance becomes simpler as there are fewer devices to oversee.

Skype, which originally marketed itself as a service among friends, has begun to cater to businesses, providing free-of-charge connections between any users on the Skype network and connecting to and from ordinary PSTN telephones for a charge.

In the United States the Social Security Administration (SSA) is converting its field offices of 63,000 workers from traditional phone installations to a VoIP infrastructure carried over its existing data network.

BENEFITS

VoIP can be a benefit for reducing communication and infrastructure costs. Examples include:

§ Routing phone calls over existing data networks to avoid the need for separate voice and data networks.

§ Conference calling, IVR, call forwarding, automatic redial, and caller ID features that traditional telecommunication companies (telcos) normally charge extra for, are available free of charge from open source VoIP implementations.

VoIP can facilitate tasks and provide services that may be more difficult to implement using the PSTN. Examples include:

§ The ability to transmit more than one telephone call over a single broadband connection.

§ Secure calls using standardized protocols (such as Secure Real-time Transport Protocol). Most of the difficulties of creating a secure telephone connection over traditional phone lines, such as digitizing and digital transmission, are already in place with VoIP. It is only necessary to encrypt and authenticate the existing data stream.

§ Location independence. Only a sufficiently fast and stable Internet connection is needed to get a connection from anywhere to a VoIP provider.

§ Integration with other services available over the Internet, including video conversation, message or data file exchange during the conversation, audio conferencing, managing address books, and passing information about whether other people are available to interested parties.

§ Unified Communications, the integration of VoIP with other business systems including E-mail, Customer Relationship Management (CRM), and Web systems.

BROADBAND CABLE MODEMS AND DSL

2012-03-11T12:58:00.003-07:00

Broadband technology basically consists of :

1.CABLE MODEMS

2.DSL

Advantages of Broadband Connections:

1. Broadband Internet service is “Always-on”

You don’t have to dial in your ISP’s server, just lunch your browser and you’re there.

2. Broadband Internet service is faster

Cable and DSL connections generally deliver an average min of around 150 Kbps and max 1~2 Mbps, versus the max of 53.3 Kbps for dial-up Internet services.

CABLE MODEMS

Cable access is the most widely available and most commonly used broadband technology in U.S.
Cable Web access is faster than dial-up and less complex to install than DSL

· Cable modems work via a 6-MHz channel slot with the cable signal to deliver burst download speeds of up to 5 Mbps

Features of Cable Modem:

Shared bandwidth
Upload speed is lower than download.
Available in residential areas, often not in downtown or commercial area

DSL

• DSL is the second –most-popular broadband service in U.S., trailing cable modem access

• DSL delivers digital data through phone lines

• DSL boasts faster connection times, high throughput, and cheaper subscription cost

IPTV TECHNOLOGY EXPLAINED

2012-03-11T12:56:00.001-07:00

Internet Protocol television (IPTV) is a system through which Internet television services are delivered using the architecture and networking methods of the Internet Protocol Suite over a packet-switched network infrastructure (such as the Internet or other access network), instead of being delivered through traditional radio frequency broadcast, satellite signal, and cable television (CATV) formats.

IPTV services may be classified into three main groups:

§ live television, with or without interactivity related to the current TV show;

§ time-shifted television: catch-up TV (replays a TV show that was broadcast hours or days ago), start-over TV (replays the current TV show from its beginning);

§ video on demand (VOD): browse a catalog of videos, not related to TV programming.

IPTV is distinguished from general Internet-based or web-based multimedia services by its on-going standardization process (e.g., European Telecommunications Standards Institute) and preferential deployment scenarios in subscriber-based telecommunications networks with high-speed access channels into end-user premises via set-top boxes or other customer-premises equipment.

IPTV ARCHITECTURE

Elements

§ TV Head-end: where live TV channels are encoded, encrypted and delivered in the form of IP multicast streams.

§ VOD platform: where on-demand video assets are stored and served when a user makes a request in the form of IP unicast stream.

§ Interactive portal: allows the user to navigate within the different IPTV services, such as the VOD catalog.

§ delivery network: the packet switched network that carries IP packets (unicast and multicast).

§ home gateway: the piece of equipment at the user's home that terminates the access link from the delivery network.

ADVANTAGES

The IP(Internet Protocol)-based platform offers significant advantages, including the ability to integrate television with other IP-based services like high speed Internet access and VoIP.

A switched IP network also allows for the delivery of significantly more content and functionality. In a typical TV or satellite network, using broadcast video technology, all the content constantly flows downstream to each customer, and the customer switches the content at the set-top box. The customer can select from as many choices as the telecomms, cable or satellite company can stuff into the “pipe” flowing into the home. A switched IP network works differently.

Content remains in the network, and only the content the customer selects is sent into the customer’s home. That frees up bandwidth, and the customer’s choice is less restricted by the size of the “pipe” into the home. This also implies that the customer's privacy could be compromised to a greater extent than is possible with traditional TV or satellite networks. It may also provide a means to hack into, or at least disrupt the private network

WHAT IS DTH??

2012-03-11T12:55:00.000-07:00

DTH stands for Direct-To-Home television. DTH is defined as the reception of satellite programmes with a personal dish in an individual home.

DTH does away with the need for the local cable operator and puts the broadcaster directly in touch with the consumer. Only cable operators can receive satellite programmes and they then distribute them to individual homes.

DTH ARCHITECTURE

A DTH network consists of a broadcasting centre, satellites, encoders, multiplexers, modulators and DTH receivers.

A DTH service provider has to lease Ku-band transponders from the satellite. The encoder converts the audio, video and data signals into the digital format and the multiplexer mixes these signals. At the user end, there will be a small dish antenna and set-top boxes to decode and view numerous channels. On the user's end, receiving dishes can be as small as 45 cm in diametre.

DTH is an encrypted transmission that travels to the consumer directly through a satellite. DTH transmission is received directly by the consumer at his end through the small dish antenna. A set-top box, unlike the regular cable connection, decodes the encrypted transmission.

How does DTH really differ from cable TV?

The way DTH reaches a consumer's home is different from the way cable TV does. In DTH, TV channels would be transmitted from the satellite to a small dish antenna mounted on the window or rooftop of the subscriber's home. So the broadcaster directly connects to the user. The middlemen like local cable operators are not there in the picture.

DTH can also reach the remotest of areas since it does away with the intermediate step of a cable operator and the wires (cables) that come from the cable operator to your house. As we explained above, in DTH signals directly come from the satellite to your DTH dish.

Also, with DTH, a user can scan nearly 700 channels!

6 REASONS FOR NOT TO BUY APPLE NEW IPAD 4

2012-03-11T12:50:00.002-07:00

So Apple held this quiet, off-the-radar shindig yesterday where they invited a few people to a tiny little iPad-related event in San Francisco, maybe you heard about it… Oh who am I kidding, you’ve probably already placed your preorder for “The new iPad” (lowercase ‘n’), as Apple officially touted it during their latest event-that-nearly-crashed-the-Internet. The new iPad is “resolutionary,” boasts the company’s website, and according to Apple executives, it has a “magical pane of glass.” If you didn’t know better, you’d think Cupertino just launched a new Harry Potter book.

But do you really need one of these “new” iPads? Maybe. They’re certainly attractive with the screen resolution doubled, all the lovely new high-res games and apps, and of course, 4G LTE support. But just in case you’re not feeling any of that “magical” love, here’s an unimpeachable haters checkli

You really, really wanted an iPad 3. Yep, Apple’s new iPad is just called “iPad,” or “The new iPad,” if you want to get technical about it. Sure, the company already does that with its MacBook lineup (ever heard of a MacBook Air 2 or a MacBook Pro 3?), but no, you definitely wanted an iPad 3, so who cares if it comes with a holodeck or lets you travel through time — you’re not spending penny one for anything without the number “3″ in the moniker.

1024 x 768 pixel resolution is “eye-popping” enough. Remember the Cyndi Lauper song “The Goonies ‘R’ Good Enough”? Insert “1024 x 768,” or just “either of the first two iPads.” Apple’s new iPad takes you higher, of course — to a glorious 2048 x 1536 pixels, or 264 ppi (pixels per inch) — but since that’s technically not iPhone 4S caliber (326 ppi), you’re not budging. “Retina” display? Pffft.

You live outside your area’s 4G LTE umbrella. Why pay upwards of $800 for an iPad that’ll only connect at prehistoric 3G speeds? What’s Apple trying to pull here? (Wait, they still sell a model that’s Wi-Fi-only for $500?)

The “new” iPad is bigger and heavier. The iPad 2 is 0.34 inches and weighs 1.33 pounds. The “new” iPad, by contrast, is 0.37 inches and weighs 1.4 pounds. Imagine the added strain on your wrists and metacarpals!

You wanted an 8-megapixel camera. The “new” iPad’s rear-facing camera only does 5-megapixels. There go all your pro photography dreams of replacing a cheaper, higher-end digital camera that fits in your pocket with a tablet that barely fits in a purse.

It doesn’t come with Siri. Sure, Apple included a voice dictation feature courtesy a mic button on the virtual keyboard, but you wanted Siri, by gosh, and if the “new” iPad won’t answer questions like “Where do babies come from?” or “How much wood would a woodchucker chuck if a woodchucker could chuck wood?” then everyone out of the pool, no “The new iPad” for you

DIFFERENCE BETWEEN MSISDN NUMBER AND IMSI

2012-03-11T12:36:00.004-07:00

IMSI - International Mobile Subscriber Identity

MSISDN - Mobile Directory Number

Both are two different numbers. One is used to identify a subscriber by operator. Other is used to route the call. A subscriber becomes customer of an operator by buying the SIM card of the operator. At this point the subscriber is allocated an IMSI and an MSISDN. IMSI is used to identify a subscriber by the operator. But MSISDN is the number which is used for dialing. So when you call your friend/a mobile, you call the MSISDN number of the mobile, not the IMSI.

For Eg: Consider 2 subscriber's A & B. And assume the values given below;

IMSI (A) : 987123111111111

IMSI (B) : 987124111111111

MSISDN (A) : 9448012121

MSISDN (B) : 9448121212

So when A would like to call B, A dials the MSISDN number of B, i.e, 9448121212. And same is true for vice-versa.

Now coming to need of IMSI, when a mobile is switched on, it needs to identify and authenticate the subscriber with the network. Only then the mobile will be able to use the network services. So to identify the user in this scenario, the IMSI of the subscriber is used. In fact the IMSI is burned in the SIM card, which will be used while authenticating with network. MSISDN will not be stored in the SIM.

220.227.79.138 10:18, 3 September 2007 (UTC)

A couple of important things were left out of that description. There are three identities IMEI, IMSI and MSISDN. The IMEI is the mobile phone's identity which is burned into the phone. The IMSI is the identity of the SIM card. And MSISDN is the phone number. You can change the phone number of the phone without changing its IMSI (the sim card). You can keep the same phone number (MSISDN), change phones (the IMEI) and just move the SIM card (IMSI). By separating the three identities, it makes it easy for subscribers to easily upgrade phones without having to change phone numbers or accounting information. Kgrr (talk) 23:57, 2 June 2008 (UTC)

3G Networks By Sumit Kasera, Nishit Narang

2012-03-10T12:31:00.001-08:00

http://books.google.co.in/books?id=b5d0_au-z9MC&pg=PA95&dq=3g+protocol+stack&hl=en&sa=X&ei=OQdbT-ruN8PMrQfCk9yiDA&ved=0CFoQ6AEwBQ#v=onepage&q&f=false

ATM FOR CUPCAKES INVENTED

2012-03-09T02:48:00.002-08:00

http://techland.time.com/2012/03/01/technology-finally-gets-useful-cupcake-atm-opens-in-los-angeles/

The Cell Phone Jammer on the Bus Goes Jam, Jam,

2012-03-09T01:43:00.001-08:00

http://techland.time.com/2012/03/02/the-cell-phone-jammer-on-the-bus-goes-jam-jam-jam/

Verizon Launches Wireless Broadband for Homes

2012-03-09T01:39:00.002-08:00

http://techland.time.com/2012/03/06/verizon-launches-wireless-broadband-for-homes/

Hands-on with the Huawei MediaPad 10 FHD

2012-03-09T01:22:00.000-08:00

http://phandroid.com/2012/02/29/hands-on-with-the-huawei-mediapad-10-fhd/